Re: [asterisk-users] Selecting a specific line from Zap/g
ignorpat is your friend On 9/30/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Sep 30, 2007 at 02:34:01AM -0700, bilal ghayyad wrote: Dear List; How can I place a call via Zap/g1 (group) but need to determine the line (FXO port) that will go via it? Simply don't use groups. Use channels directly. To dial via the specific Zaptel channel NN, use Zap/NN Am I missing anything? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX client for windows ce pda
Hi I'm looking for a iax client that will run on my htc tytn (windows ce) .. -- Greg ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Odd one way RTP on SIP to SIP calls
Hi everyone, I'm having an odd problem with one way RTP on SIP to SIP calls. I have two SIP servers, one is an Asterisk and the remote SIP server is a Nortel SIP server. When a call comes to the Nortel server through the PSTN and is routed to the Asterisk, audio is fine. Two way RTP and no problems. When a SIP client registered on the Nortel server calls the Asterisk, the Asterisk doesn't seem to send any RTP. As far as I can tell, there isn't anything wrong with the call setup. show core version shows: Asterisk 1.4.4 built by mark @ d620 on a i686 running Linux on 2007-05-17 06:39:34 UTC SIP and RTP debugging on Asterisk shows this: http://www.arnarson.net/~orn/calldebug.txt On a Trixbox Asterisk server I have at hand (Asterisk 1.2.18 built by root @ build.trixbox.org on a i686 running Linux on 2007-04-25 19:59:21 UTC) on the same network (same subnet and physical location) as the 1.4.4 this problem does not exist. There is no RTP problem when SIP clients registered on Nortel call. If anyone could help or suggest anything it would be greatly appreciated. Best regards, Örn ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What's the deal with ATAcomm?
Chiming in here, I had to return a Polycom to VOIPSupply and the turnaround time was basically immediate with no questions asked. They've always done us right here. OTH, I did have a bad glitch with ATAComm and it took a while for them to resolve the issue. That's horrible. I don't buy too many IP phones these days, but can anyone suggest a place better than the scumbags at VoIP supply? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ODBC version
What version of ODBC does asterisk 1.4 need? -- - Chris Stinson Network Operations Center ISDN-Net, Inc. 615-221-4200 x103 [EMAIL PROTECTED] ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd one way RTP on SIP to SIP calls
Is this a SIP connection or a SIP-T one? Not sure (don't have access to my previous life docs :-), but this seems to be a Session Server Trunks doing SIP-T, not sure is the configuration you want...Have you tried to contact their support ? PS: this c: application/ISUP;version=ANSI88;base=ANSI88, don't remember seeing in plain SIP calls, so that is why I suspect is configured as a SIP-T. Örn Arnarson wrote: Hi everyone, I'm having an odd problem with one way RTP on SIP to SIP calls. I have two SIP servers, one is an Asterisk and the remote SIP server is a Nortel SIP server. When a call comes to the Nortel server through the PSTN and is routed to the Asterisk, audio is fine. Two way RTP and no problems. When a SIP client registered on the Nortel server calls the Asterisk, the Asterisk doesn't seem to send any RTP. As far as I can tell, there isn't anything wrong with the call setup. show core version shows: Asterisk 1.4.4 built by mark @ d620 on a i686 running Linux on 2007-05-17 06:39:34 UTC SIP and RTP debugging on Asterisk shows this: http://www.arnarson.net/~orn/calldebug.txt On a Trixbox Asterisk server I have at hand (Asterisk 1.2.18 built by root @ build.trixbox.org on a i686 running Linux on 2007-04-25 19:59:21 UTC) on the same network (same subnet and physical location) as the 1.4.4 this problem does not exist. There is no RTP problem when SIP clients registered on Nortel call. If anyone could help or suggest anything it would be greatly appreciated. Best regards, Örn ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Voicemail
Hi I've configured my asterisk and voicemail all works fine but I want to restrict call time to voicemail that is when user calls voicemail he can use voicemail system only for a max of 5 min that is after five minutes asterisk should disconnect the call. thanks Arun ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange problem with latest Asterisk
Hi all, I'm having a problem with latest version of Asterisk. When I put someone on hold or if I dial an extension with music on hold the call hangs up after a few seconds when MUOH has changed file to play. Any thoughts? Many thanks, Christian ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd one way RTP on SIP to SIP calls
You are right, the remote server is a SIP-T. I haven't had any problems connecting it to regular SIP servers thusfar though. Also like I mentioned, I don't have this one-way RTP problem with an earlier version of Asterisk. Thanks for your reply, Örn On 10/1/07, Julio Arruda [EMAIL PROTECTED] wrote: Is this a SIP connection or a SIP-T one? Not sure (don't have access to my previous life docs :-), but this seems to be a Session Server Trunks doing SIP-T, not sure is the configuration you want...Have you tried to contact their support ? PS: this c: application/ISUP;version=ANSI88;base=ANSI88, don't remember seeing in plain SIP calls, so that is why I suspect is configured as a SIP-T. Örn Arnarson wrote: Hi everyone, I'm having an odd problem with one way RTP on SIP to SIP calls. I have two SIP servers, one is an Asterisk and the remote SIP server is a Nortel SIP server. When a call comes to the Nortel server through the PSTN and is routed to the Asterisk, audio is fine. Two way RTP and no problems. When a SIP client registered on the Nortel server calls the Asterisk, the Asterisk doesn't seem to send any RTP. As far as I can tell, there isn't anything wrong with the call setup. show core version shows: Asterisk 1.4.4 built by mark @ d620 on a i686 running Linux on 2007-05-17 06:39:34 UTC SIP and RTP debugging on Asterisk shows this: http://www.arnarson.net/~orn/calldebug.txt On a Trixbox Asterisk server I have at hand (Asterisk 1.2.18 built by root @ build.trixbox.org on a i686 running Linux on 2007-04-25 19:59:21 UTC) on the same network (same subnet and physical location) as the 1.4.4 this problem does not exist. There is no RTP problem when SIP clients registered on Nortel call. If anyone could help or suggest anything it would be greatly appreciated. Best regards, Örn ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7940G licensing with asterisk
I do not believe that the web page referenced below states that you need a license to use Cisco phones with any pbx other than Call Manager. It only states that you are required to have a license regardless of the protocol used and their documentation is specifically aimed at Call Manager implementations. Any one else have another view and or supporting document on this? Glenn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peder @ NetworkOblivion Sent: Thursday, September 27, 2007 4:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7940G licensing with asterisk I'm pretty sure that any Cisco switch that has PoE supports pre-standard PoE. However there are only certain ones that do support the standard. If you are looking for the cheapest used ones, then a 3524-PWR will work. If you want new, then a 3560 ps version will work. Erick Perez wrote: Peder, can you point me to the Cisco PoE swith (pre-802.3af) that can handle the 7940G ? The 7941G does conform to the standard but it only support SCCP (shame on cisco). On 9/27/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote: Yes, you need to buy a license if you use it with ANY pbx, whether it is Callmangler or Asterisk or whatever. If you buy one used, then you need to pay to re-license it as well. The 7940/7960 only work with Cisco PoE, not standard 802.3af, so you will need a switch that provides Cisco PoE for it to work. Erick Perez wrote: Hi there, In Cisco web site http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sh eet09186a008008884a.html It says that regardless of the technology used you have to buy a licencse. Does the license apply to use the phone with asterisk, or, can i just buy the phone? Also, the phone does not requiere to use an AC adapter if used with PoE injectors/switches. Can non-Cisco PoE injectors/switches be used with this phone? Thanks, ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd one way RTP on SIP to SIP calls
Julio, It seems you had something going there; I disallowed ISUP messages on the SIP-T server and now I have two way audio. Thanks a lot for your help! Best regards, Örn On 10/1/07, Örn Arnarson [EMAIL PROTECTED] wrote: You are right, the remote server is a SIP-T. I haven't had any problems connecting it to regular SIP servers thusfar though. Also like I mentioned, I don't have this one-way RTP problem with an earlier version of Asterisk. Thanks for your reply, Örn On 10/1/07, Julio Arruda [EMAIL PROTECTED] wrote: Is this a SIP connection or a SIP-T one? Not sure (don't have access to my previous life docs :-), but this seems to be a Session Server Trunks doing SIP-T, not sure is the configuration you want...Have you tried to contact their support ? PS: this c: application/ISUP;version=ANSI88;base=ANSI88, don't remember seeing in plain SIP calls, so that is why I suspect is configured as a SIP-T. Örn Arnarson wrote: Hi everyone, I'm having an odd problem with one way RTP on SIP to SIP calls. I have two SIP servers, one is an Asterisk and the remote SIP server is a Nortel SIP server. When a call comes to the Nortel server through the PSTN and is routed to the Asterisk, audio is fine. Two way RTP and no problems. When a SIP client registered on the Nortel server calls the Asterisk, the Asterisk doesn't seem to send any RTP. As far as I can tell, there isn't anything wrong with the call setup. show core version shows: Asterisk 1.4.4 built by mark @ d620 on a i686 running Linux on 2007-05-17 06:39:34 UTC SIP and RTP debugging on Asterisk shows this: http://www.arnarson.net/~orn/calldebug.txt On a Trixbox Asterisk server I have at hand (Asterisk 1.2.18 built by root @ build.trixbox.org on a i686 running Linux on 2007-04-25 19:59:21 UTC) on the same network (same subnet and physical location) as the 1.4.4 this problem does not exist. There is no RTP problem when SIP clients registered on Nortel call. If anyone could help or suggest anything it would be greatly appreciated. Best regards, Örn ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Changing contexts on the fly
Hi,
Many thanks all for the useful tips - I've gone with a (simple!) mySQL table
with a flag in it, indicating the day/night mode, adding the following into
the dialplan:
[external]
; other stuff in here, excluded for clarity
; Include the SJS phone line controls
include = sjs_ctrl
[sjs_ctrl]
; Determine if we're in or out of the office, and divert accordingly
; Note - callerID is set because it doesn't get it from the line :(
exten = s,1,NoOp(-- ${CALLERID(number)} calling on ZAP channel)
exten = s,2,Set(CALLERID(number)=unknown)
exten = s,3,Set(CALLERID(name)=SJS Line 1)
exten = s,4,MYSQL(Connect connid db_server login_id super_secret_password
db_name)
exten = s,5,MYSQL(Query resultid ${connid} SELECT\ currentStatus\ FROM\
myStatus)
exten = s,6,MYSQL(Fetch fetchid ${resultid} MyStatus)
exten = s,7,MYSQL(Clear ${resultid})
exten = s,8,MYSQL(Disconnect ${connid})
exten = s,9,GotoIf($[${MyStatus} = y]?10:12)
exten = s,10,GoTo(sjs,s,1)
exten = s,12,Goto(sjs-ooh,s,1)
[sjs]
exten = s,1,NoOp(-- ${CALLERID(number)} calling on ZAP channel)
exten = s,n,Set(CALLERID(number)=unknown)
exten = s,n,Set(CALLERID(name)=SJS Line 1)
exten = s,n,Dial(SIP/5100,30)
exten = s,n,Answer()
exten = s,n,Wait(0.75)
exten = s,n,Voicemail(5100,u)
exten = s,n,Hangup()
[sjs-ooh]
exten = s,1,Answer()
exten = s,n,Wait(0.75)
exten = s,n,Playback(thank-you-for-calling [etc - lots more soundfiles
here])
exten = s,n,Voicemail(5100,s)
exten = s,n,Hangup()
Then, in the internal extensions config, I've added the following:
; Switch SJS day/night modes
;Daytime (star star D)
exten = **3,1,NoCdr()
exten = **3,n,Answer()
exten = **3,n,MYSQL(Connect connid db_server login_id super_secret_password
db_name)
exten = **3,n,MYSQL(Query resultid ${connid} UPDATE\ myStatus\ SET\
currentStatus\ = \ \'n\')
exten = **3,n,MYSQL(Clear ${resultid})
exten = **3,n,MYSQL(Disconnect ${connid})
exten = **3,n,Playback(daytime)
exten = **3,n,Hangup()
;Nighttime (star star N)
exten = **6,1,NoCDR()
exten = **6,n,Answer()
exten = **6,n,MYSQL(Connect connid db_server login_id super_secret_password
db_name)
exten = **6,n,MYSQL(Query resultid ${connid} UPDATE\ myStatus\ SET\
currentStatus\ = \ \'n\')
exten = **6,n,MYSQL(Clear ${resultid})
exten = **6,n,MYSQL(Disconnect ${connid})
exten = **6,n,Playback(nighttime)
exten = **6,n,Hangup()
So I can switch between day night modes with **D or **N (3 or 6
respectively) :) Dead simple stuff so far, I may get more whizzy with it
later on... At some point, I'll probably switch over to a fully realtime
config, so I can DIY my own user interface.
Cheers!
Ade.
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Re: [asterisk-users] Cisco 7940G licensing with asterisk
In trying to verify licensing requirements I called Tech-Data and spoke to the Cisco licensing reps there (my company is set up as a reseller through Tech-Data) and was informed by them that a license for Cisco VoIP phones is only required if connecting it to a Call Manager or any other Cisco voice technology solution such as a Cisco router. If you are connecting a Cisco phone to any other pbx they consider it a third party solution and licensing requirements for that vendor are your responsibility. Glenn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peder @ NetworkOblivion Sent: Thursday, September 27, 2007 12:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7940G licensing with asterisk Yes, you need to buy a license if you use it with ANY pbx, whether it is Callmangler or Asterisk or whatever. If you buy one used, then you need to pay to re-license it as well. The 7940/7960 only work with Cisco PoE, not standard 802.3af, so you will need a switch that provides Cisco PoE for it to work. Erick Perez wrote: Hi there, In Cisco web site http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186 a008008884a.html It says that regardless of the technology used you have to buy a licencse. Does the license apply to use the phone with asterisk, or, can i just buy the phone? Also, the phone does not requiere to use an AC adapter if used with PoE injectors/switches. Can non-Cisco PoE injectors/switches be used with this phone? Thanks, ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What's the deal with ATAcomm?
Andrew Joakimsen wrote: That's horrible. I don't buy too many IP phones these days, but can anyone suggest a place better than the scumbags at VoIP supply? http://www.pcp.ch/ or http://www.digitec.ch/ /Per Jessen, Zürich ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unauthorized 401
Hi, I'm trying to register SIP phone with an asterisk serve, failing miserably. The server is sending 401 Unauthorized responses to the registration attempts, but every time the phone is re-REGISTERing without authorization. I'd think this was a problem with the IP phone, except... the very same phone registers correctly (authenticated) with another asterisk box, same brand, similarly configured. The phone is a Leadtek BVP 8882 videophone. The bad asterisk server has the following build info, but I haven't seen any bug reports for this problem... Linux aadk 2.6.16.27sx00i-1.0.3.1 #2 Thu Aug 30 13:18:42 CDT 2007 blackfin unknown Asterisk Build: Asterisk autotag_for_sx00i-1.0.3 (sx00i 1.0.3.1) Asterisk GUI-version Revision: 1453 $ I'm wondering if the 401 unauthorized response has bad formatting. I compared the bad asterisk server repeated response, with the good asterisk server first response (the phone includes authorization in subsequent REGISTER for that one). The only difference I can see, is that the bad asterisk responses have a blank Access-URL: line before WWW-Authenticate. I've included log from the bad asterisk server. If necessary I can provide one from the good server as well, but I've left it out for now to avoid confusion. Asterisk Business Edition autotag_for_sx00i-1.0.3 (sx00i 1.0.3.1), Copyright (C) 1999 - 2007 Digium, Inc. and others. Created by Mark Spencer Thank you for using Business Edition. This Software is provided by Digium Inc under license. Please refer to the license agreement provided with the Software. === Connected to Asterisk autotag_for_sx00i-1.0.3 (sx00i 1.0.3.1) currently running on aadk (pid = 304) aadk*CLI sip debug aadk*CLI SIP Debugging enabled [Kaadk*CLI The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead. [Kaadk*CLI core set debug 255 aadk*CLI Core debug was 0 and is now 255 [Kaadk*CLI core set verbose 255 aadk*CLI Verbosity was 0 and is now 255 [Kaadk*CLI --- SIP read from 192.168.220.31:5060 --- REGISTER sip:asterisk.foo.internal SIP/2.0 Call-ID: [EMAIL PROTECTED] From: 6001sip:[EMAIL PROTECTED];tag=10007c00-4bc9 To: 6001sip:[EMAIL PROTECTED] CSeq: 101 REGISTER Via: SIP/2.0/UDP 192.168.220.31:5060;branch=z9hG4bKc0a8efe310007f004bce Contact: sip:[EMAIL PROTECTED]:5060 Max-Forwards: 70 User-Agent: LRSTD LR8882 2.5.00_99 Expires: 300 Content-Length: 0 - --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.220.31 : 5060 (no NAT) --- Transmitting (no NAT) to 192.168.220.31:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.220.31:5060;branch=z9hG4bKc0a8efe310007f004bce;received=192.168.220.31 From: 6001sip:[EMAIL PROTECTED];tag=10007c00-4bc9 To: 6001sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] Access-URL: Content-Length: 0 --- Transmitting (no NAT) to 192.168.220.31:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.220.31:5060;branch=z9hG4bKc0a8efe310007f004bce;received=192.168.220.31 From: 6001sip:[EMAIL PROTECTED];tag=10007c00-4bc9 To: 6001sip:[EMAIL PROTECTED];tag=as1aa11ae2 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Access-URL: WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=141ab0a6 Content-Length: 0 [Kaadk*CLI Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: REGISTER) [Kaadk*CLI --- SIP read from 192.168.220.31:5060 --- REGISTER sip:asterisk.foo.internal SIP/2.0 Call-ID: [EMAIL PROTECTED] From: 6001sip:[EMAIL PROTECTED];tag=10007c00-4bc9 To: 6001sip:[EMAIL PROTECTED] CSeq: 101 REGISTER Via: SIP/2.0/UDP 192.168.220.31:5060;branch=z9hG4bKc0a8efe310007f004bce Contact: sip:[EMAIL PROTECTED]:5060 Max-Forwards: 70 User-Agent: LRSTD LR8882 2.5.00_99 Expires: 300 Content-Length: 0 - --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.220.31 : 5060 (no NAT) --- Transmitting (no NAT) to 192.168.220.31:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.220.31:5060;branch=z9hG4bKc0a8efe310007f004bce;received=192.168.220.31 From: 6001sip:[EMAIL PROTECTED];tag=10007c00-4bc9 To: 6001sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] Access-URL: Content-Length: 0 --- Transmitting (no NAT) to 192.168.220.31:5060 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.220.31:5060;branch=z9hG4bKc0a8efe310007f004bce;received=192.168.220.31 From: 6001sip:[EMAIL
Re: [asterisk-users] Odd one way RTP on SIP to SIP calls
Just a guess in fact..but.. I'm sure others would love to know how is the NGSS (SST now ?) config for this purpose, as well as your sip.conf and etc (one note, you are running SN09 or ISN09 ? Not sure, but this also would help others out there.. :-) Örn Arnarson wrote: Julio, It seems you had something going there; I disallowed ISUP messages on the SIP-T server and now I have two way audio. Thanks a lot for your help! Best regards, Örn On 10/1/07, Örn Arnarson [EMAIL PROTECTED] wrote: You are right, the remote server is a SIP-T. I haven't had any problems connecting it to regular SIP servers thusfar though. Also like I mentioned, I don't have this one-way RTP problem with an earlier version of Asterisk. Thanks for your reply, Örn On 10/1/07, Julio Arruda [EMAIL PROTECTED] wrote: Is this a SIP connection or a SIP-T one? Not sure (don't have access to my previous life docs :-), but this seems to be a Session Server Trunks doing SIP-T, not sure is the configuration you want...Have you tried to contact their support ? PS: this c: application/ISUP;version=ANSI88;base=ANSI88, don't remember seeing in plain SIP calls, so that is why I suspect is configured as a SIP-T. Örn Arnarson wrote: Hi everyone, I'm having an odd problem with one way RTP on SIP to SIP calls. I have two SIP servers, one is an Asterisk and the remote SIP server is a Nortel SIP server. When a call comes to the Nortel server through the PSTN and is routed to the Asterisk, audio is fine. Two way RTP and no problems. When a SIP client registered on the Nortel server calls the Asterisk, the Asterisk doesn't seem to send any RTP. As far as I can tell, there isn't anything wrong with the call setup. show core version shows: Asterisk 1.4.4 built by mark @ d620 on a i686 running Linux on 2007-05-17 06:39:34 UTC SIP and RTP debugging on Asterisk shows this: http://www.arnarson.net/~orn/calldebug.txt On a Trixbox Asterisk server I have at hand (Asterisk 1.2.18 built by root @ build.trixbox.org on a i686 running Linux on 2007-04-25 19:59:21 UTC) on the same network (same subnet and physical location) as the 1.4.4 this problem does not exist. There is no RTP problem when SIP clients registered on Nortel call. If anyone could help or suggest anything it would be greatly appreciated. Best regards, Örn ___ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Voicemail
Hi Arun - I've configured my asterisk and voicemail all works fine but I want to restrict call time to voicemail that is when user calls voicemail he can use voicemail system only for a max of 5 min that is after five minutes asterisk should disconnect the call. Do you mean that you want the maximum message length to be 5 minutes? If so, you can use maxmessage in the general section of voicemail.conf. It's set in seconds, so: [general] maxmessage=300 - Noah ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What's the deal with ATAcomm?
Vahan Yerkanian wrote: Andrew Kohlsmith wrote: On Saturday 29 September 2007 18:43:59 Andrew Joakimsen wrote: That's horrible. I don't buy too many IP phones these days, but can anyone suggest a place better than the scumbags at VoIP supply? I don't know about you, but I've had nothing but very good results with VOIPSupply. I didnt do huge business with them, but I have purchased new and refurb polycoms from them without so much as an ounce of pain. -A. I've bought more than $10k worth of equipment from voipsupply.com across the globe and they've always treated me very professionally. All their shipments always arrived on time and were well packed and documented. Just my 2 cents, Vahan Yes, I have to concur. Good to fair prices, good customer service, fast and professional shipping. Their only weakness seems to be the RMA process and that might be up to snuff by now. It has been several months since I was stuck trying to RMA something. They even admitted that their RMA process needed some work, now that is honesty. Thanks, Steve Totaro ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What's the deal with ATAcomm?
On 9/30/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote: I don't know about you, but I've had nothing but very good results with VOIPSupply. I didnt do huge business with them, but I have purchased new and refurb polycoms from them without so much as an ounce of pain. Ditto - I've never had a single problem with them. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Park problem on IAX2 channel
Hi all, I have 2 asterisk box connected with IAX trunk. One box have connected a SIP phone and the second have a TDM card with one analog phone. When from SIP phone I try to park the call from analog phone with #700 the call is correctly parked but in the second asterisk I see this log: -- Executing Dial(Zap/2-1, IAX2/CTM1/STI1|30|rjtT) -- Called CTM1/STI1 -- Call accepted by 172.16.4.1 (format alaw) -- Format for call is alaw -- IAX2/CTM1-2 answered Zap/2-1 -- Started music on hold, class 'default', on IAX2/CTM1-2 -- Zap/2-1 Playing 'pbx-transfer' (language 'en') -- Unable to find extension '77' in context 'from-internal' -- Zap/2-1 Playing 'pbx-invalid' (language 'en') -- Stopped music on hold on IAX2/CTM1-2 The line: -- Unable to find extension '77' in context 'from-internal' appears also with '#', '#7', '', '0'... It seems that the dtmf came across the iax channel and arrive to other asterisk. The are a way to block this dtmf across the IAX trunk? Thanks Enrico. -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web: http://www.pasqualotto.org http://www.linkedin.com/in/epasqualotto smime.p7s Description: S/MIME Cryptographic Signature ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 423 Interval Too Brief back from
I am having problems with SIP Registration. There has been an article about the issue (http://www.asteriskguru.com/archives/asterisk-users-sip-registration-problem-w-sbc-vt96867.html ) but I am not able to apply the patch. I am using AsteriskNow beta6. The message I am having is: [Oct 1 19:28:54] NOTICE[19102]: chan_sip.c:7247 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #2) -- parse_srv: SRV mapped to host mgc.voip.server, port 5060 -- Got SIP response 423 Interval Too Brief back from voip_server_IP asteriskNow*CLI In AsteriskNow beta I was able to apply the patch and I solved the problem. Any idea? Thanks, Nejc ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd one way RTP on SIP to SIP calls
Good point. Here goes. I am running ISN09 (recently upgraded). Actually the upgrade caused a lot of problems and now the CS2K has to be datafilled so that the Asterisk trunks are Q764 and not Q767, lest the calls fail. Additionally the NGSS/SST had to be patched up to date to fix another issue. The NGSS config is pretty straight forward, no fancy options set. In this version of * I had to change the following options to make it work with this version of Asterisk: Use OPTIONS for Heartbeat: No Enforce CODEC-Compatibility: No (oddly enough, as the codecs are compatible) Accepts Encapsulated ISUP: No sip.conf entry is like this: [Nortel-SIP] type=friend host=1.1.1.1 port=5060 dtmfmode=rfc2833 canreinvite=no disallow=all allow=alaw allow=ulaw context=default I think most of the other options were left at default, even though I don't think that they are crucial. Best regards, Örn On 10/1/07, Julio Arruda [EMAIL PROTECTED] wrote: Just a guess in fact..but.. I'm sure others would love to know how is the NGSS (SST now ?) config for this purpose, as well as your sip.conf and etc (one note, you are running SN09 or ISN09 ? Not sure, but this also would help others out there.. :-) Örn Arnarson wrote: Julio, It seems you had something going there; I disallowed ISUP messages on the SIP-T server and now I have two way audio. Thanks a lot for your help! Best regards, Örn On 10/1/07, Örn Arnarson [EMAIL PROTECTED] wrote: You are right, the remote server is a SIP-T. I haven't had any problems connecting it to regular SIP servers thusfar though. Also like I mentioned, I don't have this one-way RTP problem with an earlier version of Asterisk. Thanks for your reply, Örn On 10/1/07, Julio Arruda [EMAIL PROTECTED] wrote: Is this a SIP connection or a SIP-T one? Not sure (don't have access to my previous life docs :-), but this seems to be a Session Server Trunks doing SIP-T, not sure is the configuration you want...Have you tried to contact their support ? PS: this c: application/ISUP;version=ANSI88;base=ANSI88, don't remember seeing in plain SIP calls, so that is why I suspect is configured as a SIP-T. Örn Arnarson wrote: Hi everyone, I'm having an odd problem with one way RTP on SIP to SIP calls. I have two SIP servers, one is an Asterisk and the remote SIP server is a Nortel SIP server. When a call comes to the Nortel server through the PSTN and is routed to the Asterisk, audio is fine. Two way RTP and no problems. When a SIP client registered on the Nortel server calls the Asterisk, the Asterisk doesn't seem to send any RTP. As far as I can tell, there isn't anything wrong with the call setup. show core version shows: Asterisk 1.4.4 built by mark @ d620 on a i686 running Linux on 2007-05-17 06:39:34 UTC SIP and RTP debugging on Asterisk shows this: http://www.arnarson.net/~orn/calldebug.txt On a Trixbox Asterisk server I have at hand (Asterisk 1.2.18 built by root @ build.trixbox.org on a i686 running Linux on 2007-04-25 19:59:21 UTC) on the same network (same subnet and physical location) as the 1.4.4 this problem does not exist. There is no RTP problem when SIP clients registered on Nortel call. If anyone could help or suggest anything it would be greatly appreciated. Best regards, Örn ___ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd one way RTP on SIP to SIP calls
Sorry for the spam, but there was a typo. I was running ISN09, but the upgrade was to ISN09u, which I am currently running. That was the upgrade that caused the interoperability problem with Asterisk that I mentioned. On 10/1/07, Örn Arnarson [EMAIL PROTECTED] wrote: Good point. Here goes. I am running ISN09 (recently upgraded). Actually the upgrade caused a lot of problems and now the CS2K has to be datafilled so that the Asterisk trunks are Q764 and not Q767, lest the calls fail. Additionally the NGSS/SST had to be patched up to date to fix another issue. The NGSS config is pretty straight forward, no fancy options set. In this version of * I had to change the following options to make it work with this version of Asterisk: Use OPTIONS for Heartbeat: No Enforce CODEC-Compatibility: No (oddly enough, as the codecs are compatible) Accepts Encapsulated ISUP: No sip.conf entry is like this: [Nortel-SIP] type=friend host=1.1.1.1 port=5060 dtmfmode=rfc2833 canreinvite=no disallow=all allow=alaw allow=ulaw context=default I think most of the other options were left at default, even though I don't think that they are crucial. Best regards, Örn On 10/1/07, Julio Arruda [EMAIL PROTECTED] wrote: Just a guess in fact..but.. I'm sure others would love to know how is the NGSS (SST now ?) config for this purpose, as well as your sip.conf and etc (one note, you are running SN09 or ISN09 ? Not sure, but this also would help others out there.. :-) Örn Arnarson wrote: Julio, It seems you had something going there; I disallowed ISUP messages on the SIP-T server and now I have two way audio. Thanks a lot for your help! Best regards, Örn On 10/1/07, Örn Arnarson [EMAIL PROTECTED] wrote: You are right, the remote server is a SIP-T. I haven't had any problems connecting it to regular SIP servers thusfar though. Also like I mentioned, I don't have this one-way RTP problem with an earlier version of Asterisk. Thanks for your reply, Örn On 10/1/07, Julio Arruda [EMAIL PROTECTED] wrote: Is this a SIP connection or a SIP-T one? Not sure (don't have access to my previous life docs :-), but this seems to be a Session Server Trunks doing SIP-T, not sure is the configuration you want...Have you tried to contact their support ? PS: this c: application/ISUP;version=ANSI88;base=ANSI88, don't remember seeing in plain SIP calls, so that is why I suspect is configured as a SIP-T. Örn Arnarson wrote: Hi everyone, I'm having an odd problem with one way RTP on SIP to SIP calls. I have two SIP servers, one is an Asterisk and the remote SIP server is a Nortel SIP server. When a call comes to the Nortel server through the PSTN and is routed to the Asterisk, audio is fine. Two way RTP and no problems. When a SIP client registered on the Nortel server calls the Asterisk, the Asterisk doesn't seem to send any RTP. As far as I can tell, there isn't anything wrong with the call setup. show core version shows: Asterisk 1.4.4 built by mark @ d620 on a i686 running Linux on 2007-05-17 06:39:34 UTC SIP and RTP debugging on Asterisk shows this: http://www.arnarson.net/~orn/calldebug.txt On a Trixbox Asterisk server I have at hand (Asterisk 1.2.18 built by root @ build.trixbox.org on a i686 running Linux on 2007-04-25 19:59:21 UTC) on the same network (same subnet and physical location) as the 1.4.4 this problem does not exist. There is no RTP problem when SIP clients registered on Nortel call. If anyone could help or suggest anything it would be greatly appreciated. Best regards, Örn ___ ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Selecting a specific line from Zap/g
ignorepat continues dialtone after a leading digit has been dialed on FXS ports. How does ignorepat help this guy? Al lists wrote: ignorpat is your friend On 9/30/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Sep 30, 2007 at 02:34:01AM -0700, bilal ghayyad wrote: Dear List; How can I place a call via Zap/g1 (group) but need to determine the line (FXO port) that will go via it? Simply don't use groups. Use channels directly. To dial via the specific Zaptel channel NN, use Zap/NN Am I missing anything? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7940G licensing with asterisk
Hi, also I have called Cisco suport to ask how to use SIP protocol on Cisco 7941G (and my Astersik), the their answer is the following: ..SIP Firmware for the 7941G phone only works with Call Manager 5.x. You must have CCM 5.x to use this firmware, is needeful to buy a CCM license for use SIP protocol Asterisk. -- Salvatore. - Original Message - From: Glenn Cobb [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Monday, October 01, 2007 4:21 PM Subject: Re: [asterisk-users] Cisco 7940G licensing with asterisk In trying to verify licensing requirements I called Tech-Data and spoke to the Cisco licensing reps there (my company is set up as a reseller through Tech-Data) and was informed by them that a license for Cisco VoIP phones is only required if connecting it to a Call Manager or any other Cisco voice technology solution such as a Cisco router. If you are connecting a Cisco phone to any other pbx they consider it a third party solution and licensing requirements for that vendor are your responsibility. Glenn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peder @ NetworkOblivion Sent: Thursday, September 27, 2007 12:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7940G licensing with asterisk Yes, you need to buy a license if you use it with ANY pbx, whether it is Callmangler or Asterisk or whatever. If you buy one used, then you need to pay to re-license it as well. The 7940/7960 only work with Cisco PoE, not standard 802.3af, so you will need a switch that provides Cisco PoE for it to work. Erick Perez wrote: Hi there, In Cisco web site http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186 a008008884a.html It says that regardless of the technology used you have to buy a licencse. Does the license apply to use the phone with asterisk, or, can i just buy the phone? Also, the phone does not requiere to use an AC adapter if used with PoE injectors/switches. Can non-Cisco PoE injectors/switches be used with this phone? Thanks, ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which Asterisk version to use?
On Sun, 2007-09-30 at 10:49 -0400, Eric B. wrote: Thanks for the advice everyone. Will continue reading TFOT and get started! For what it's worth, the second edition of Asterisk: The Future of Telephony is now available as a free PDF from http://openbooks.oreilly.com/. (It's obviously available in book form from your favorite bookseller as well.) -Jared ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7940G licensing with asterisk
I just got SIP firmware images from Cisco for installation on 7970G. Cisco requires you buy a SmartNet account (about $15, no other dependencies apply) that entitles you to download a SIP firmware image file from their protected support website. The 7970G now needs a different image than the other 79xx phones, but the same rules apply to all of them. Those rules do not require any other license or other restriction, once you have legitimately obtained and installed the firmware on the phone, to use the phones with Asterisk (or any other 3rd party system). Of course, to use the phones with Cisco's CallManager product, you must have a licensed copy of the CallManager product, with all the other restrictions and fees that come with it. FWIW, the procedure of buying that SIP image from Cisco was a nightmare. I had to buy the SmartNet account from a reseller which did nothing to ensure that I completed the download transaction that was the stated purpose (as they described it to me) of buying the license. Then navigating to the license I needed, among the many versions and revisions, was confusing and opaque. The SmartNet account took days to send to me, and didn't work for the required access when it arrived. Cisco consumed an entire workweek to deliver the license that didn't unlock the website, then of course ignored requests for support through the weekend (into which their late delivery forced my request to be made). When I finally got Cisco to respond, they did deliver a knowledgeable and honest support tech who stuck with me until I had everything I needed to proceed. Though every stated maximum turnaround time for every phase in the process was exceeded, sometimes by many multiples. But since the image can be used only with a Cisco phone, which must (ultimately) be bought from Cisco, the kafkaesque procedure is intolerable. The image should be a one-click download that charges your credit card and comes with a SmartNet account, if they absolutely must charge the $15. In a sane world, the SIP image wouldn't have any restrictions, a free download that people could just email each other (or its URL), because its distribution would market Cisco phones. But probably Cisco knows that the SIP image lets (free) Asterisk compete with its proprietary CallManager, so they make it both a revenue source, and as complicated as possible. On Mon, 2007-10-01 at 09:43 -0500, [EMAIL PROTECTED] wrote: Message: 18 Date: Mon, 1 Oct 2007 10:21:34 -0400 From: Glenn Cobb [EMAIL PROTECTED] Subject: Re: [asterisk-users] Cisco 7940G licensing with asterisk To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII In trying to verify licensing requirements I called Tech-Data and spoke to the Cisco licensing reps there (my company is set up as a reseller through Tech-Data) and was informed by them that a license for Cisco VoIP phones is only required if connecting it to a Call Manager or any other Cisco voice technology solution such as a Cisco router. If you are connecting a Cisco phone to any other pbx they consider it a third party solution and licensing requirements for that vendor are your responsibility. Glenn -- (C) Matthew Rubenstein ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7940G licensing with asterisk
Matthew Rubenstein wrote: I just got SIP firmware images from Cisco for installation on 7970G. Cisco requires you buy a SmartNet account (about $15, no other dependencies apply) that entitles you to download a SIP firmware image file from their protected support website. The 7970G now needs a different image than the other 79xx phones, but the same rules apply to all of them. Those rules do not require any other license or other restriction, once you have legitimately obtained and installed the firmware on the phone, to use the phones with Asterisk (or any other 3rd party system). Of course, to use the phones with Cisco's CallManager product, you must have a licensed copy of the CallManager product, with all the other restrictions and fees that come with it. FWIW, the procedure of buying that SIP image from Cisco was a nightmare. I had to buy the SmartNet account from a reseller which did nothing to ensure that I completed the download transaction that was the stated purpose (as they described it to me) of buying the license. Then navigating to the license I needed, among the many versions and revisions, was confusing and opaque. The SmartNet account took days to send to me, and didn't work for the required access when it arrived. Cisco consumed an entire workweek to deliver the license that didn't unlock the website, then of course ignored requests for support through the weekend (into which their late delivery forced my request to be made). When I finally got Cisco to respond, they did deliver a knowledgeable and honest support tech who stuck with me until I had everything I needed to proceed. Though every stated maximum turnaround time for every phase in the process was exceeded, sometimes by many multiples. But since the image can be used only with a Cisco phone, which must (ultimately) be bought from Cisco, the kafkaesque procedure is intolerable. The image should be a one-click download that charges your credit card and comes with a SmartNet account, if they absolutely must charge the $15. In a sane world, the SIP image wouldn't have any restrictions, a free download that people could just email each other (or its URL), because its distribution would market Cisco phones. But probably Cisco knows that the SIP image lets (free) Asterisk compete with its proprietary CallManager, so they make it both a revenue source, and as complicated as possible. The way I understand it, that $15 doesn't actually even give you the right to use the SIP firmware. It only gives you the right to access the download area. The whole model is silly, at best. -- Jason Parker Digium ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7940G licensing with asterisk
I was told 7941G were sold with SIP firmware factory installed. Does anyone know this to be true or not ? Regards ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7940G licensing with asterisk
Matthew, Did you keep any hardcopy of licensing terms (when downloading SIP firmware) ? This way we might double check if CCM license is mandatory to connect a Cisco SIP phone to an Asterisk server. Beside that, Cisco SIP phones require menu localization files to come from CCM. Did you run into this ? Is there anything special with these phones that make those localization files to be downloaded (I know that's another topic, but while we're at it ...) Regards ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] When is a new release with this DTMF patch going to come out?
http://bugs.digium.com/view.php?id=10535 It is quite serious, costing us money and ill will from our customers. Yes, we are still running 1.2. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7940G licensing with asterisk
On Mon, 2007-10-01 at 11:44 -0500, Jason Parker wrote: Matthew Rubenstein wrote: I just got SIP firmware images from Cisco for installation on 7970G. The way I understand it, that $15 doesn't actually even give you the right to use the SIP firmware. It only gives you the right to access the download area. The whole model is silly, at best. When I explained to each of the account reseller and the Cisco support that I was going to use the SIP firmware to connect to Asterisk, not CallManager, they each told me only that Cisco wouldn't support (trouble tickets and other tech support time) the system using Asterisk, though they did explicitly assure me (as does the documentation) that since the SIP firmware is RFC-compliant, it would work with any RFC-compliant server, not just CallManager (and so would work with SIP RFC-complaint Asterisk). It's a giant game of CYA. I spent hours getting my $15 worth from the SIP download. I'm surprised a bitter backlash hasn't made these SIP images widely available for download around the Web. I think they might have the serial# of the phone they're registered to when the account is created, and of course the contract states otherwise, but I'd still expect Cisco's deliberately difficult process hasn't created enemies who'd do it anyway. Maybe there are just so few people using it this way that none have materialized (yet). So I guess Cisco's PITA plan is working. -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which Asterisk version to use?
Eric, It's a huge learning curve, but you'll soon see light at ahead even before you know a lot. Get the book and start playing. You won't be sorry! On 9/30/07, Eric B. [EMAIL PROTECTED] wrote: Jim Canfield [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Eric B. wrote: site and got to chapter 4 or 5 and decided to take a break. Which is when I found AsteriskNow and TriBox and then started wondering if it was really necessary / worthwhile to figure out all the intricacies of the application if someones have already created the appliance version of it. In which case, I was very confused as to the difference btwn AsteriskNow and TriBox. Thanks! Last week I posed a similar question to the list as a noob. Specifically, I was curious why every one was so adverse to GUI implementations. Like you, I entered the asterisk world quite idealistic and oblivious to what is actually required to create a functional system (still am). I spent the good part of last week trying to make heads or tails of the AsteriskNOW distro, but finally gave up in favor of a plain jane Debian install with asterisk and wish I would have never wasted so much time trying to figure out how the users.conf worked. quote [TK]D-Fender - The users.conf is a flaming piece of sh**! \quote I actually thought that was a bit harsh when I read it...turns out to be quite accurate. Long story short, I'm learning to be quite comfortable in the CLI and finding myself more productive in nano (yes..nano) than I was in the GUI. Good luck! Thanks for the advice everyone. Will continue reading TFOT and get started! Eric ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ODBC version for cdr?
I'm having an error when I try to ./configure asterisk using --with-odbc=/usr/lib. Below is the version of each application and the ./configure below that. Any help would be appreciated. unixODBC-2.2.11-7.1 unixODBC-devel-2.2.11-7.1 mysql-connector-odbc-3.51.12-2.2 mysql-5.0.22-2.1 Contents of odbcinst.ini # Driver from the MyODBC package # Setup from the unixODBC package [MySQL] Description = ODBC for MySQL Driver = /usr/lib/libmyodbc.so Setup = /usr/lib/libodbcmyS.so FileUsage = 1 checking for SQLConnect in -lodbc... no configure: *** configure: *** The unixODBC installation on this system appears to be broken. configure: *** Either correct the installation, or run configure configure: *** without explicitly specifying --with-odbc -- - Chris Stinson Network Operations Center ISDN-Net, Inc. 615-221-4200 x103 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7940G licensing with asterisk
On Mon, 2007-10-01 at 19:02 +0200, Olivier wrote: Matthew, Did you keep any hardcopy of licensing terms (when downloading SIP firmware) ? This way we might double check if CCM license is mandatory to connect a Cisco SIP phone to an Asterisk server. I haven't seen any such mandate, and didn't elicit one when I told Cisco I was using the firmware/phones with Asterisk instead of CallManager. I don't think there is one. You can look at the release notes for all the 7900 firmware available for download, including the version I got: http://www.cisco.com/en/US/products/hw/phones/ps379/prod_release_notes_list.html . Beside that, Cisco SIP phones require menu localization files to come from CCM. Did you run into this ? Is there anything special with these phones that make those localization files to be downloaded (I know that's another topic, but while we're at it ...) I have not completed the deployment of the phones, as I've had other priorities. I have not yet run into that problem, or heard of it before, but it might be lying in wait later in the process. I'd like to know whether it is indeed a problem in using the phones with Asterisk, and how to solve it if so. Regards -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What's the deal with ATAcomm?
You should probably post that question on the Asterisk business forum. -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/ -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andrew Joakimsen Sent: Saturday, September 29, 2007 3:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What's the deal with ATAcomm? That's horrible. I don't buy too many IP phones these days, but can anyone suggest a place better than the scumbags at VoIP supply? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How To Transfer Asterisk Installation to a Different Machine
I am having some hardware problems with the Linux machine where I have Asterisk installed and want to use a different machine. Assuming I install Asterisk on machine number 2, is it possible to just move files over from the old machine to the new machine and the new machine will behave like the old? Anyone have a list of the files that would need to be moved? (Obviously the *.conf files in the Asterisk directory, I can think of some others, but if someone ever did a list that would be a great help.) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ODBC version for cdr?
On Mon, 2007-10-01 at 12:52 -0500, Chris Stinson wrote: I'm having an error when I try to ./configure asterisk using --with-odbc=/usr/lib. Below is the version of each application and the ./configure below that. Any help would be appreciated. The autoconf magic in Asterisk looks for a shared library provided by the libtool-ltdl package (at least under Red Hat, CentOS, and Fedora), and won't detect the ODBC libraries without it. (Yes, the build system *should* be a little more informative about this.) -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ODBC version for cdr?
The libtool-ltdl package is installed. On 10/1/07, Jared Smith [EMAIL PROTECTED] wrote: On Mon, 2007-10-01 at 12:52 -0500, Chris Stinson wrote: I'm having an error when I try to ./configure asterisk using --with-odbc=/usr/lib. Below is the version of each application and the ./configure below that. Any help would be appreciated. The autoconf magic in Asterisk looks for a shared library provided by the libtool-ltdl package (at least under Red Hat, CentOS, and Fedora), and won't detect the ODBC libraries without it. (Yes, the build system *should* be a little more informative about this.) -- Jared Smith Community Relations Manager Digium, Inc. -- - Chris Stinson Network Operations Center ISDN-Net, Inc. 615-221-4200 x103 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How To Transfer Asterisk Installation to a Different Machine
On 10/1/07, Robert DeVries [EMAIL PROTECTED] wrote: Anyone have a list of the files that would need to be moved? (Obviously the *.conf files in the Asterisk directory, I can think of some others, but if someone ever did a list that would be a great help.) You'll probably want to move the subdirs of /var/spool/asterisk that apply to your install as well. -Erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ODBC version for cdr?
I believe libtool-ltdl-devel is what you need. On Mon, 2007-10-01 at 13:22 -0500, Chris Stinson wrote: The libtool-ltdl package is installed. On 10/1/07, Jared Smith [EMAIL PROTECTED] wrote: On Mon, 2007-10-01 at 12:52 -0500, Chris Stinson wrote: I'm having an error when I try to ./configure asterisk using --with-odbc=/usr/lib. Below is the version of each application and the ./configure below that. Any help would be appreciated. The autoconf magic in Asterisk looks for a shared library provided by the libtool-ltdl package (at least under Red Hat, CentOS, and Fedora), and won't detect the ODBC libraries without it. (Yes, the build system *should* be a little more informative about this.) -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How To Transfer Asterisk Installation to a Different Machine
I'm not sure I did it right, but I always just moved the following: /etc/asterisk/*.conf /var/spool/asterisk /var/lib/asterisk /usr/lib/asterisk (may be unnecessary; only for non-typical modules; see below) And I haven't had any problems, assuming all required modules are in the new /usr/lib/asterisk. If any are missing, for example codec_g729a.so, the missing ones can be grabbed from /usr/lib/asterisk on the old system. Moj Robert DeVries wrote: I am having some hardware problems with the Linux machine where I have Asterisk installed and want to use a different machine. Assuming I install Asterisk on machine number 2, is it possible to just move files over from the old machine to the new machine and the new machine will behave like the old? Anyone have a list of the files that would need to be moved? (Obviously the *.conf files in the Asterisk directory, I can think of some others, but if someone ever did a list that would be a great help.) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ODBC version for cdr?
If this is on a RedHat-type system (EL, Fedora, but also CentOS), make sure you have a symlink in place for libltdl.so. Even though I also had the libtool-ltdl package installed, it only provided libs and links for /usr/lib/libltdl.so..3.1.4 and libltdl.so.3. It did not create a symlink to a plain-jane libltdl.so library, which is what was needed here to successfully ./configure. On 10/1/07, Chris Stinson [EMAIL PROTECTED] wrote: The libtool-ltdl package is installed. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which Asterisk version to use?
Razza wrote: On 27/09/2007, Eric B. [EMAIL PROTECTED] wrote: For starters, what is the difference btwn the 1.2 and 1.4 branches of Asterisk? I can't seem to find a document that describes the changes. Anyone? Not much/Lots Depends what you're looking for. Important considerations for us in moving to 1.4 were: jabber/gtalk support t.38 passthrough support shared line appearance support You can probably have a look at the Changelogs for more details. If you don't need the extra features 1.2.Current is still the most stable solution IMO. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ODBC version for cdr?
I didn't have libtool-ltdl-devel. Once I install the devel package, it finished the configuration. Thanks James, Jared and Kai-Uwe for the responses. On 10/1/07, Kai-Uwe Jensen [EMAIL PROTECTED] wrote: If this is on a RedHat-type system (EL, Fedora, but also CentOS), make sure you have a symlink in place for libltdl.so. Even though I also had the libtool-ltdl package installed, it only provided libs and links for /usr/lib/libltdl.so..3.1.4 and libltdl.so.3. It did not create a symlink to a plain-jane libltdl.so library, which is what was needed here to successfully ./configure. On 10/1/07, Chris Stinson [EMAIL PROTECTED] wrote: The libtool-ltdl package is installed. -- - Chris Stinson Network Operations Center ISDN-Net, Inc. 615-221-4200 x103 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When is a new release with this DTMF patch going to come out?
Unfortunately 1.2 is no longer getting bug fixes (except for security fixes). You will have to manually apply the patch for 1.2. Yes the 1.2 maint policy sucks for many people, including me. Doug wrote: http://bugs.digium.com/view.php?id=10535 It is quite serious, costing us money and ill will from our customers. Yes, we are still running 1.2. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Voicemail
Do you mean, when people call VoiceMailMain to _check_ their messages they need to be cut off after five minutes? For this, I'd put an absolute timeout before the call to VoiceMailMain. I'm using asterisk 1.4, and the following syntax works for me: ; Set absolute timeout to five minutes (300 seconds) exten = 777,1,Set(TIMEOUT(absolute)=300) exten = 777,2,VoiceMailMain Moj Arun Kumar wrote: Hi I've configured my asterisk and voicemail all works fine but I want to restrict call time to voicemail that is when user calls voicemail he can use voicemail system only for a max of 5 min that is after five minutes asterisk should disconnect the call. thanks Arun ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme conference using g729?
In my experience, and theoretically by design, it doesn't matter what codec you are using when you call a meetme conference. Moj Mark Quitoriano wrote: Hi, is there a way to use g729 in meetme? Thanks! ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When is a new release with this DTMF patch going to come out?
At 14:14 10/1/2007, Eric \ManxPower\ Wieling wrote: Unfortunately 1.2 is no longer getting bug fixes (except for security fixes). You will have to manually apply the patch for 1.2. Yes the 1.2 maint policy sucks for many people, including me. Hmmm. Many people believe that 1.4 is still quite buggy. (Yes, some are actually using it on production servers.) I suspect that probably over 80% of Asterisk servers are running 1.2. By not releasing bug fixes in a new 1.2 release it seems that there is quite a bit of ill will being created. Doug wrote: http://bugs.digium.com/view.php?id=10535 It is quite serious, costing us money and ill will from our customers. Yes, we are still running 1.2. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unauthorized 401
Jason Kincaid [EMAIL PROTECTED] writes: Hi, I'm trying to register SIP phone with an asterisk serve, failing miserably. The server is sending 401 Unauthorized responses to the registration attempts, but every time the phone is re-REGISTERing without authorization. I'd think this was a problem with the IP phone, except... the very same phone registers correctly (authenticated) with another asterisk box, same brand, similarly configured. --- Transmitting (no NAT) to 192.168.220.31:5060 --- Is it possible that the Asterisk server is trying to send to a NAT IP which it can't actually reach? -- Kyle Sexton ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Voicemail
In article [EMAIL PROTECTED], Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: Do you mean, when people call VoiceMailMain to _check_ their messages they need to be cut off after five minutes? For this, I'd put an absolute timeout before the call to VoiceMailMain. I'm using asterisk 1.4, and the following syntax works for me: ; Set absolute timeout to five minutes (300 seconds) exten = 777,1,Set(TIMEOUT(absolute)=300) exten = 777,2,VoiceMailMain It would also be nice to the user if you define the T extension to play an announcment to the user (you have reached your time limit) before hanging up, so they don't just think the system broke. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unauthorized 401
I have both units on my desk here, the server is on the local 224 subnet and the phone is on 220 subnet (IP 192.168.220.31). My PC is on the same jack as the phone, sharing a hub, so I can sniff packets with ethereal. My PC can see the 401 unauthorized packets so therefore the phone can too. -Original Message- From: Kyle Sexton [mailto:[EMAIL PROTECTED] Sent: Monday, October 01, 2007 2:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Jason Kincaid Subject: Re: [asterisk-users] Unauthorized 401 Jason Kincaid [EMAIL PROTECTED] writes: Hi, I'm trying to register SIP phone with an asterisk serve, failing miserably. The server is sending 401 Unauthorized responses to the registration attempts, but every time the phone is re-REGISTERing without authorization. I'd think this was a problem with the IP phone, except... the very same phone registers correctly (authenticated) with another asterisk box, same brand, similarly configured. --- Transmitting (no NAT) to 192.168.220.31:5060 --- Is it possible that the Asterisk server is trying to send to a NAT IP which it can't actually reach? -- Kyle Sexton ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Tor3e on x86_64
Hi list, Have somebody tried a tor3e board on a intel x86_64 ? I have installed one but I have no audio on it, but, installing on a x86 32 bits server it works fine. I'm using asterisk-1.4.11 and zaptel-tor3-1.4.5.1.tar.gz. Ard ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk+Sipura 3102+PSTN line
Hello Gurus I've installed my Asterisk server for testing on the company I work the setup or the approach let's call it is: 1 Asterisk Server fully configured and with some SIP extensions setup on two cities A and B. 2. One local PSTN line connected thru a x01p card to call local phone numbers numbres on city A. 3. A Sipura 3102 Gateway on city B connected to a city's B PSTN line. I wnat to be able to call from city A to city B PSTN phone numbers from city A using Internet and vice-versa. What is the proper config on Asterisk and the SPA-3102 so that I can call SIP extension on that device plus PSTN phone lines. Thanks for the tips or the pages/guides I can be referred to. Thanks! :) -- DAVID GONZALEZ H. GNU/Linux Debian+SuSE+RedHat+LFS TECNICO EN REDES NETWORK ADMIN http://www.computrabajo.com.co/cvs/dgonzalezh ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When is a new release with this DTMF patch going to come out?
Doug wrote: At 14:14 10/1/2007, Eric \ManxPower\ Wieling wrote: Unfortunately 1.2 is no longer getting bug fixes (except for security fixes). You will have to manually apply the patch for 1.2. Yes the 1.2 maint policy sucks for many people, including me. Hmmm. Many people believe that 1.4 is still quite buggy. (Yes, some are actually using it on production servers.) I suspect that probably over 80% of Asterisk servers are running 1.2. By not releasing bug fixes in a new 1.2 release it seems that there is quite a bit of ill will being created. I agree with you. I do NOT work for Digium. The Digium official statement (as far as I can tell) can be seen at: http://lists.digium.com/pipermail/asterisk-security/2007-August/000186.html ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When is a new release with this DTMF patch going to come out?
Eric ManxPower Wieling wrote: Doug wrote: At 14:14 10/1/2007, Eric \ManxPower\ Wieling wrote: Unfortunately 1.2 is no longer getting bug fixes (except for security fixes). You will have to manually apply the patch for 1.2. Yes the 1.2 maint policy sucks for many people, including me. Hmmm. Many people believe that 1.4 is still quite buggy. (Yes, some are actually using it on production servers.) I suspect that probably over 80% of Asterisk servers are running 1.2. By not releasing bug fixes in a new 1.2 release it seems that there is quite a bit of ill will being created. I agree with you. I do NOT work for Digium. The Digium official statement (as far as I can tell) can be seen at: http://lists.digium.com/pipermail/asterisk-security/2007-August/000186.html I call for a 1.2 spoon or fork! Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7940G licensing with asterisk
My understanding is: Smartnet: service contract basically allows you to download the newest sw release. Besides that you can buy phones without a license. Presumably as spares But you must buy a SIP license to technically be allowed to use that software that can be obtained from Smartnet. I know there was some changes a year or two back, but wasn't that just pricing? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7940G licensing with asterisk
Just buy the Linksys SPA962's they work better than the cisco phones in a NAT env. /b On Oct 1, 2007, at 6:13 PM, Andrew Joakimsen wrote: My understanding is: Smartnet: service contract basically allows you to download the newest sw release. Besides that you can buy phones without a license. Presumably as spares But you must buy a SIP license to technically be allowed to use that software that can be obtained from Smartnet. I know there was some changes a year or two back, but wasn't that just pricing? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme conference using g729?
but is there a way to use g729 codec in meetme? On 10/2/07, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: In my experience, and theoretically by design, it doesn't matter what codec you are using when you call a meetme conference. Moj ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mISDN NPI setting with b410p
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 For the archives, the quick fix was to change: p[2] = 0x80 + (type4) + plan; to: p[2] = 0x80; Problem now resolved and system working well. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHAYR7DQNt8rg0Kp4RArcAAJ9/Mt1OjDtp+NQSQk8NLJ6RW0f0QwCbBTqA Jops5j7yUw5rr0NC7fj2LUg= =795H -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP trought Firewall
Hi to everyone! I have succerfully instaled my new Asterisk 1.4 on my debian etch. I have my users in sip.conf like this: [200] type=peer host=dynamic context=home secret=200 callerid= 200 dtmfmode=rfc2833 nat=yes [EMAIL PROTECTED] disallow=all allow=ulaw I can make calls in my LAN but i can´t ear comunications with another client trought Internet. My Asterisk is in my LAN and i not have a DMZ. I search in the list and find something about rtp == rtp.conf. I found rtpstart and rtpend and forward those Ports on my firewall, but this don´t work for me. What´s wrong??? If you need some info please tell me. Thanks in advance! Emiliano Vazquez. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme conference using g729?
Ok Let me chime in on this one. If you can use ulaw/alaw because you'll end up with tandem encoding which will make the conference sound worse to some people. All audio coming in will get transcoded to signed linear and pushed down into zaptel then back up and out to the conference participants. You'll end up with the best audio quality if you limit the transcoding. /b On Oct 1, 2007, at 6:37 PM, Mark Quitoriano wrote: but is there a way to use g729 codec in meetme? On 10/2/07, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: In my experience, and theoretically by design, it doesn't matter what codec you are using when you call a meetme conference. Moj ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme conference using g729?
As long as you have some g729 codecs installed, Asterisk will do this fine. PaulH On Tue, 2007-10-02 at 07:37 +0800, Mark Quitoriano wrote: but is there a way to use g729 codec in meetme? On 10/2/07, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: In my experience, and theoretically by design, it doesn't matter what codec you are using when you call a meetme conference. Moj ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI Setup problem
Hi everyone, I'm trying to get a Sangoma A101D-X card talking to a Sasktel PRI (Megalink) circuit and having some trouble getting it to handshake. Thanks for any help or suggestions to diagnose this problem. The problem is that Asterisk has trouble bringing up the link. I see the following lines every couple of minutes: == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up h87*CLI pri show spans PRI span 1/0: Provisioned, Up, Active h87*CLI pri show spans PRI span 1/0: Provisioned, Up, Active == Primary D-Channel on span 1 down [Oct 1 17:52:49] WARNING[13164]: chan_zap.c:2393 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! h87*CLI pri show spans PRI span 1/0: Provisioned, Down, Active == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up h87*CLI pri show spans PRI span 1/0: Provisioned, Up, Active == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 down [Oct 1 17:55:20] WARNING[13164]: chan_zap.c:2393 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! h87*CLI pri show spans PRI span 1/0: Provisioned, Down, Active Of course I cannot dial out: -- Executing [EMAIL PROTECTED]:1] NoOp(SIP/368-081f51d8, Dial Time of Day via PRI) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/368-081f51d8, ZAP/3|2446411|30|Tt) in new stack [Oct 1 18:01:27] WARNING[13623]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'ZAP' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED] :3] Congestion(SIP/368-081f51d8, ) in new stack == Spawn extension (default, 2446411, 3) exited non-zero on 'SIP/368-081f51d8' If I turn on pri debugging, I see lots of: h87*CLI pri debug span 1 Enabled debugging on span 1 Sending Set Asynchronous Balanced Mode Extended Sending Set Asynchronous Balanced Mode Extended Sending Set Asynchronous Balanced Mode Extended Periodically I see: Sending Set Asynchronous Balanced Mode Extended Sending Set Asynchronous Balanced Mode Extended -- Got SABME from network peer. Sending Unnumbered Acknowledgement q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED == Primary D-Channel on span 1 up -- Got SABME from network peer. Sending Unnumbered Acknowledgement q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED == Primary D-Channel on span 1 up -- Got SABME from network peer. Sending Unnumbered Acknowledgement q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED == Primary D-Channel on span 1 up -- Got SABME from network peer. Sending Unnumbered Acknowledgement q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED == Primary D-Channel on span 1 up Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 0/0x0) (Originator) Message type: RESTART (70) [18 03 a9 83 88] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 8 ] [79 01 80] Restart Indentifier (len= 3) [ Ext: 1 Spare: 0 Resetting Indicated Channel (0) ] -- Timeout occured, restarting PRI q921.c:356 t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED Sending Set Asynchronous Balanced Mode Extended q921.c:150 q921_send_sabme: q921_state now is Q921_AWAITING_ESTABLISH == Primary D-Channel on span 1 down [Oct 1 18:04:09] WARNING[13164]: chan_zap.c:2393 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! Sending Set Asynchronous Balanced Mode Extended Sending Set Asynchronous Balanced Mode Extended Any help is greatly appreciated! Thanks, Alvin ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Setup problem
The only time I have had this problem is when there was a version mismatch between Zaptel and Asterisk. Once I resolved that issue (latest asterisk + latest zaptel + reasonably recent wanpipe) everything worked for me. Alvin Austin wrote: Hi everyone, I'm trying to get a Sangoma A101D-X card talking to a Sasktel PRI (Megalink) circuit and having some trouble getting it to handshake. Thanks for any help or suggestions to diagnose this problem. The problem is that Asterisk has trouble bringing up the link. I see the following lines every couple of minutes: == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up h87*CLI pri show spans PRI span 1/0: Provisioned, Up, Active h87*CLI pri show spans PRI span 1/0: Provisioned, Up, Active == Primary D-Channel on span 1 down [Oct 1 17:52:49] WARNING[13164]: chan_zap.c:2393 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! h87*CLI pri show spans PRI span 1/0: Provisioned, Down, Active == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up h87*CLI pri show spans PRI span 1/0: Provisioned, Up, Active == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 down [Oct 1 17:55:20] WARNING[13164]: chan_zap.c:2393 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! h87*CLI pri show spans PRI span 1/0: Provisioned, Down, Active Of course I cannot dial out: -- Executing [EMAIL PROTECTED]:1] NoOp(SIP/368-081f51d8, Dial Time of Day via PRI) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/368-081f51d8, ZAP/3|2446411|30|Tt) in new stack [Oct 1 18:01:27] WARNING[13623]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'ZAP' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED] :3] Congestion(SIP/368-081f51d8, ) in new stack == Spawn extension (default, 2446411, 3) exited non-zero on 'SIP/368-081f51d8' If I turn on pri debugging, I see lots of: h87*CLI pri debug span 1 Enabled debugging on span 1 Sending Set Asynchronous Balanced Mode Extended Sending Set Asynchronous Balanced Mode Extended Sending Set Asynchronous Balanced Mode Extended Periodically I see: Sending Set Asynchronous Balanced Mode Extended Sending Set Asynchronous Balanced Mode Extended -- Got SABME from network peer. Sending Unnumbered Acknowledgement q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED == Primary D-Channel on span 1 up -- Got SABME from network peer. Sending Unnumbered Acknowledgement q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED == Primary D-Channel on span 1 up -- Got SABME from network peer. Sending Unnumbered Acknowledgement q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED == Primary D-Channel on span 1 up -- Got SABME from network peer. Sending Unnumbered Acknowledgement q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED == Primary D-Channel on span 1 up Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 0/0x0) (Originator) Message type: RESTART (70) [18 03 a9 83 88] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 8 ] [79 01 80] Restart Indentifier (len= 3) [ Ext: 1 Spare: 0 Resetting Indicated Channel (0) ] -- Timeout occured, restarting PRI q921.c:356 t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED Sending Set Asynchronous Balanced Mode Extended q921.c:150 q921_send_sabme: q921_state now is Q921_AWAITING_ESTABLISH == Primary D-Channel on span 1 down [Oct 1 18:04:09] WARNING[13164]: chan_zap.c:2393 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! Sending Set Asynchronous Balanced Mode Extended Sending Set Asynchronous Balanced Mode Extended Any help is greatly appreciated! Thanks, Alvin ___ --Bandwidth and Colocation Provided by
Re: [asterisk-users] SIP trought Firewall
Hi suffered that issue since I started that´s the course oif all of us newbies, noone is willing to help/and even answer, I don't even know if my messages are being read on this list cause not evena google for it i've received. I'm now acroos the rive with that problem you're being the victim of an unconfigured sip_nat.conf, in thre you have to specify you public static ip or dynamic domain name, your internal lan like 192.168.1.0/255.255.255.0 and natt=yes restart asterisk and you're problem will be solved, here's my config, fell free to copy/paste it but remember to change the lines to suit you're setup. nat=yes ; key line externip=my.dynamic.hot.tld; This is your addres externrefresh=30 ; some refresh time, still don't know what it does :-P localnet=192.168.1.0/255.255.255.0 ; this is the LAN setup, these are the adress range that you're DHCP NAT device is giving you. qualify=yes ; I hope you're at least sneaked-peaked Asterisk TFOT and know what qualify mean. Try this out and you'll be very happy that people outside your lan will hear you and you will hear them too. Thanks for using Asterisk and though supporting OSS the good people that developed it. On 10/1/07, Emiliano Vazquez [EMAIL PROTECTED] wrote: Hi to everyone! I have succerfully instaled my new Asterisk 1.4 on my debian etch. I have my users in sip.conf like this: [200] type=peer host=dynamic context=home secret=200 callerid= 200 dtmfmode=rfc2833 nat=yes [EMAIL PROTECTED] disallow=all allow=ulaw I can make calls in my LAN but i can´t ear comunications with another client trought Internet. My Asterisk is in my LAN and i not have a DMZ. I search in the list and find something about rtp == rtp.conf. I found rtpstart and rtpend and forward those Ports on my firewall, but this don´t work for me. What´s wrong??? If you need some info please tell me. Thanks in advance! Emiliano Vazquez. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- DAVID GONZALEZ H. GNU/Linux Debian+SuSE+RedHat+LFS TECNICO EN REDES NETWORK ADMIN http://www.computrabajo.com.co/cvs/dgonzalezh ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Setup problem
As soon as I saw channel '24 as D-channel' my guess is that the card/config is set up as T1, when you need E1. PaulH On Mon, 2007-10-01 at 18:23 -0600, Alvin Austin wrote: Hi everyone, I'm trying to get a Sangoma A101D-X card talking to a Sasktel PRI (Megalink) circuit and having some trouble getting it to handshake. Thanks for any help or suggestions to diagnose this problem. The problem is that Asterisk has trouble bringing up the link. I see the following lines every couple of minutes: == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up h87*CLI pri show spans PRI span 1/0: Provisioned, Up, Active h87*CLI pri show spans PRI span 1/0: Provisioned, Up, Active == Primary D-Channel on span 1 down [Oct 1 17:52:49] WARNING[13164]: chan_zap.c:2393 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! h87*CLI pri show spans PRI span 1/0: Provisioned, Down, Active == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up h87*CLI pri show spans PRI span 1/0: Provisioned, Up, Active == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 down [Oct 1 17:55:20] WARNING[13164]: chan_zap.c:2393 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! h87*CLI pri show spans PRI span 1/0: Provisioned, Down, Active Of course I cannot dial out: -- Executing [EMAIL PROTECTED]:1] NoOp(SIP/368-081f51d8, Dial Time of Day via PRI) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/368-081f51d8, ZAP/3|2446411|30|Tt) in new stack [Oct 1 18:01:27] WARNING[13623]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'ZAP' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED] :3] Congestion(SIP/368-081f51d8, ) in new stack == Spawn extension (default, 2446411, 3) exited non-zero on 'SIP/368-081f51d8' If I turn on pri debugging, I see lots of: h87*CLI pri debug span 1 Enabled debugging on span 1 Sending Set Asynchronous Balanced Mode Extended Sending Set Asynchronous Balanced Mode Extended Sending Set Asynchronous Balanced Mode Extended Periodically I see: Sending Set Asynchronous Balanced Mode Extended Sending Set Asynchronous Balanced Mode Extended -- Got SABME from network peer. Sending Unnumbered Acknowledgement q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED == Primary D-Channel on span 1 up -- Got SABME from network peer. Sending Unnumbered Acknowledgement q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED == Primary D-Channel on span 1 up -- Got SABME from network peer. Sending Unnumbered Acknowledgement q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED == Primary D-Channel on span 1 up -- Got SABME from network peer. Sending Unnumbered Acknowledgement q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED == Primary D-Channel on span 1 up Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 0/0x0) (Originator) Message type: RESTART (70) [18 03 a9 83 88] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 8 ] [79 01 80] Restart Indentifier (len= 3) [ Ext: 1 Spare: 0 Resetting Indicated Channel (0) ] -- Timeout occured, restarting PRI q921.c:356 t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED Sending Set Asynchronous Balanced Mode Extended q921.c:150 q921_send_sabme: q921_state now is Q921_AWAITING_ESTABLISH == Primary D-Channel on span 1 down [Oct 1 18:04:09] WARNING[13164]: chan_zap.c:2393 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! Sending Set Asynchronous Balanced Mode Extended Sending Set Asynchronous Balanced Mode Extended Any help is greatly appreciated! Thanks, Alvin ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE
Re: [asterisk-users] meetme conference using g729?
Hello Mark, On 10/2/07, Mark Quitoriano [EMAIL PROTECTED] wrote: but is there a way to use g729 codec in meetme? You have to buy a G.729 license for each channel which I believe is at USD 10.00 if I'm not mistaken. Then, make sure that your machine is fast enough for transcoding. But the best solution that I think is a codec passthrough which I think is not supported in Asterisk. Good luck! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme conference using g729?
Since the channels have to be mixed together by Asterisk, passthrough can't be supported in this case. In other circumstances, passthru works fine. PaulH On Tue, 2007-10-02 at 10:02 +0800, GNUbie wrote: Hello Mark, On 10/2/07, Mark Quitoriano [EMAIL PROTECTED] wrote: but is there a way to use g729 codec in meetme? You have to buy a G.729 license for each channel which I believe is at USD 10.00 if I'm not mistaken. Then, make sure that your machine is fast enough for transcoding. But the best solution that I think is a codec passthrough which I think is not supported in Asterisk. Good luck! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Setup problem
I've recompiled with the latest svn sources for zaptel, libpri, and Asterisk. Wanpipe is 3.3.0.p4. Switched the T1 cable. Same result. (It's a Sasktel Megalink T1/PRI circuit) CLI shows: ~~ == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up == Primary D-Channel on span 1 down [Oct 1 20:15:19] WARNING[7120]: chan_zap.c:2393 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up CLI pri show spans alternates between: PRI span 1/0: Provisioned, Down, Active (most of the time) and: PRI span 1/0: Provisioned, Up, Active(during its retry sequence - below) with debugging enabled: CLI pri debug span 1 ~~ Sending Set Asynchronous Balanced Mode Extended [..] Sending Set Asynchronous Balanced Mode Extended -- Got SABME from network peer. Sending Unnumbered Acknowledgement q921.c:709 q921_reset: q921_state now is Q921_LINK_CONNECTION_RELEASED q921.c:664 q921_dchannel_up: q921_state now is Q921_LINK_CONNECTION_ESTABLISHED == Primary D-Channel on span 1 up [above paragraph repeated 7 more times] Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 0/0x0) (Originator) Message type: RESTART (70) [18 03 a9 83 82] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 2 ] [79 01 80] Restart Indentifier (len= 3) [ Ext: 1 Spare: 0 Resetting Indicated Channel (0) ] -- Timeout occured, restarting PRI q921.c:356 t200_expire: q921_state now is Q921_LINK_CONNECTION_RELEASED Sending Set Asynchronous Balanced Mode Extended q921.c:150 q921_send_sabme: q921_state now is Q921_AWAITING_ESTABLISH == Primary D-Channel on span 1 down [Oct 1 20:30:49] WARNING[7003]: chan_zap.c:2393 pri_find_dchan: No D-channels available! Using Primary channel 24 as D-channel anyway! Sending Set Asynchronous Balanced Mode Extended [..] Sending Set Asynchronous Balanced Mode Extended # wanrouter status ~~ Devices currently active: wanpipe1 Wanpipe Config: Device name | Protocol Map | Adapter | IRQ | Slot/IO | If's | CLK | Baud rate | wanpipe1| N/A | A101/1D/A102/2D/4/4D/8| 16 | 4 | 1| EXT | 0 | Wanrouter Status: Device name | Protocol | Station | Status| wanpipe1| AFT HDLC | N/A | Connected | File /etc/zaptel.conf is: ~~ # Autogenerated by /usr/local/sbin/sangoma/setup-sangoma -- do not hand edit # Zaptel Channels Configurations (zaptel.conf) # loadzone=us defaultzone=us #Sangoma A101 port 1 [slot:4 bus:41 span: 1] span=1,0,0,esf,b8zs bchan=1-23 dchan=24 File /etc/asterisk/zapata.conf is: ~~ ;Zaptel Channels Configurations (zapata.conf) ; ;For detailed zapata options, view /etc/asterisk/zapata.conf.orig [trunkgroups] [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no ;Sangoma A101 port 1 [slot:4 bus:41 span: 1] ; I also tried: switchtype=national switchtype=dms100 context=pstn-pri group=1 signalling=pri_cpe channel = 1-23 ~~ Still struggling; thanks for any help and ideas. Alvin ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Setup problem
Alvin Austin wrote: I've recompiled with the latest svn sources for zaptel, libpri, and Asterisk. Wanpipe is 3.3.0.p4. Switched the T1 cable. Same result. Hmn -- when you recompiled, did you 1. clean out all the source directories? 2. remove the binaries? 3. recompile in the right order? I'm not sure using SVN is a good idea here. It should work with stable ;) Has the PRI been tested with test equipment? We should make sure there is a D channel before assuming misconfiguration. I don't think we can do even a loopback test if there is no D channel... -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Redundancy
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 What we do is the following: Our CPE (Customer premises equipment) registers via IAX with all of our servers at the same time (with qualify turned on for the links). All of the servers first try to reach numbers via local IAX links. If this fails they do a DUNDi lookup to the other servers to check if they are able to terminate the call. With regards to PSTN connectivity each server has a collection of methods to terminate the call with ISDN failover. Every minute each of the VoIP links are checked and their results stored in the routing table. Routes that are not accessible are temporarily removed till their responses improve. A destination is the selected based on: 1) Availability 2) Weight 3) Price The choice is made in the above order. Some providers are not very good at terminating some destinations even though the connection to them might be fine. We use this to decide on the weight. Better quality termination gets a higher weight. We then take the destinations with the highest weight (100 if the route is fine). If there are multiple destinations with the same price in this group, we chose the cheaper one. In all of the CPE the calls failover to the other servers if they are unavailable (the qualify setting does this). So, as long as there are no calls on a particular box, you can just stop Asterisk and do whatever you like. Each server updates all other server's MySQL database for credit when they are available. If not, a replication conflict email is sent so that I can manually tally any problem credits. If I thought about this properly I could probably make this automatic. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHAc0KDQNt8rg0Kp4RAn5xAJ41jLnhml3HRXj7O86ZJVPZNd2j7ACgjWXm ERH/Gj4r6j06c0LOC0/8VPQ= =QzBV -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Selecting a specific line from Zap/g
Correction, on FXO port not FXS, second, read his email first: Also, how it will be possible to assign an dedicated line (connected to FXO) to an button on the Polycom IP Phone or Broadtel IP Phone, so if user select that button then he will be sure that his outside call will be via that specific line. Just assign a key on your phone to dial that extension, and you will have dial tone on selected line, then as a traditional PBX you can send any digits to your provider. On 10/1/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: ignorepat continues dialtone after a leading digit has been dialed on FXS ports. How does ignorepat help this guy? Al lists wrote: ignorpat is your friend On 9/30/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Sep 30, 2007 at 02:34:01AM -0700, bilal ghayyad wrote: Dear List; How can I place a call via Zap/g1 (group) but need to determine the line (FXO port) that will go via it? Simply don't use groups. Use channels directly. To dial via the specific Zaptel channel NN, use Zap/NN Am I missing anything? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Panel?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Terry Giufre-Sweetser wrote: Dear List, Has anyone found or written a status panel application, windows or linux, that uses SIP notifies and subscriptions, to gather the status of SIP extensions from Asterisk? And displsy nicely on a GUI? I wrote a program a while ago - don't know if it will still work: http://www.sineapps.com/sinepeers.php Let me know if you want the sourcecode, it's probably buried somewhere in my svn repository. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHAdqJDQNt8rg0Kp4RAjbCAKCt01nH1hGq3estWpoFLeYsdypq6QCgvaHf Eeo66dSiiOKJmkqoohsoT8g= =rcag -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Supermicro PDSME+ and TE110P [ ref:00D36mPe.50033qy57:ref ] NEW CASE 22828
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Olivier wrote: Strange ! We successfully used SuperMicro boards without any IRQ problems. What is SuperMicro's reply, concerning this IRQ problems ? They sure have interest to solve this or at least explain why it can't be done. It's not the motherboard. It's the Intel e1000 network card driver. If you get a board that uses a network card supported by a different driver you won't have problems. I had one SuperMicro rackmount which had the e1000 driver and serious problems. Seeing as the machine only had one PCI slot, I couldn't add an extra network card. I moved it to another SuperMicro machine and the problems have gone. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHAdtADQNt8rg0Kp4RAun9AJ9lkiJASoJePXZI1kjp7xsqf/hGowCeOi4A TtLFXfLNDxIovXRB3txR4jY= =fFer -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager Originate Action and Cancel
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Santiago Aguiar wrote: I'm using the Originate Action on the Asterisk Manager to place calls between two extensions in async mode. Is there any way to cancel the Originate Action before I get the OriginateResponse action? I'm unable to perform a Hangup because I can't know the channel name before I get the response... I haven't seen anything that would allow for this. Have you checked the bare event output to see if you get a new channel event or something? - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHAduHDQNt8rg0Kp4RAk+JAJwKu7jX5lxD9Wi20RKJyVVI8OWK7QCgu4zX q5BwDfBGKK4Vul9XhTqYsVY= =ppPI -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
